Plugin Demo: SIP Gateway (libre)

Demo details

This demo shows how you can make use of the SIP plugin to interact with a
SIP Proxy (e.g., Kamailio) or PBX (e.g., Asterisk) in order to place or
receive calls to and from other SIP clients. Specifically, it uses the libre-based
SIP plugin: in case you're interested in the Sofia-based one, check
this other demo instead. Notice that both
plugins only exchange SIP messages from within the plugin itself: no SIP
is done in JavaScript, except for references to SIP URIs.

When started, the demo will allow you to insert a minimum set of information
required to REGISTER the web page as a SIP client at a SIP Proxy or PBX you specify.
This will allow you to call SIP URIs, or receive calls through the SIP Server itself.
During a call, you'll also be able to interact with the PBX via DTMF tones, e.g.,
to drive an Interactive Voice Response (IVR) menu that you're being presented with.

Note well! This plugin is currently WIP, and so
may not always work as expected. Considering the Sofia-based plugin has been around
for much longer, and as of today has been used by more people in production, you
may want to stick to that if what you're looking for is stability.